• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2008.05a
• /
• pp.285-288
• /
• 2008

In ubiquitous computing environment the biological signal ECG (Electrocardiogram signal) is usually recorded with noise components. Adaptive interference (or noise) canceller do adaptive filtering of the noise reference input to maximally match and subtract out noise or interference from the primary (signal plus noise) input thereby adaptively eliminate unwanted interference from the ECG signal. Measured Stress ECG (or exercise ECG signal) signal have three major noisy component like baseline wander noise, motion artifact noise and EMG (Electro-mayo-cardiogram) noise. These noises are not only distorted signal but also root of incorrect diagnosis while ECG data are analyzed. Motion artifact and EMG noises behave like wide band spectrum signals, and they considerably do overlapping with the ECG spectrum. Here the multidimensional adaptive method used for filtering which is more effective to improve signal to noise ratio.

• Journal of KIISE:Computing Practices and Letters
• /
• v.6 no.4
• /
• pp.403-410
• /
• 2000

In this paper, the development of a stock information retrieval system using speech recognition and its features are described. The system is based on DHMM (discrete hidden Markov model) and PLUs (phonelike units) are used as the basic unit for recognition. End-point detection and echo cancellation are included to facilitate speech input. Continuous speech recognizer is implemented to allow multi-word speech. Data collected over several months are analyzed.

• Proceedings of the IEEK Conference
• /
• 2004.06a
• /
• pp.55-58
• /
• 2004

This this paper, we propose vertical Bell laboratories layered space time (V-BLAST) system based on variable rate Low-Density Parity Check (LDPC) codes to improve performance of receiver when QR decomposition interference suppression combined with interference cancellation is used over independent Rayleigh fading channel. The different rate LDPC codes can be made by puncturing some rows of a given parity check matrix. This allows to implement a single encoder and decoder for different rate LDPC codes. The performance can be improved by assigning stronger LDPC codes in lower layer than upper layer because the poor SNR of first detected data streams makes error propagation. Keeping the same overall code rates, the V-BLAST system with different rate LDPC codes has the better performance (in terms of Bit Error Rate) than with constant rate LDPC code in fast fading channel.

• Proceedings of the KIEE Conference
• /
• 2008.10b
• /
• pp.113-114
• /
• 2008

In this paper, binary pulse-position modulation (2PPM) time-hoping (TH) ultra-wideband (UWB) system is applied to multiple input multiple output (MIMO) system using vertical bell lab layered space-time (V-BLAST) structure to achieve high-data-rate communications. This UWB-MIMO system and its receivers are analyzed, and its BER performances are evaluated. In the receiver, various MIMO detection algorithms such as zero-forcing (ZF), ZF-ordered successive interference cancellation (OSIC), minimum-mean-square-error (MMSE), MMSE-OSIC and maximum likelihood (ML) are comparatively studied.

• Proceedings of the KIEE Conference
• /
• 2006.10c
• /
• pp.351-353
• /
• 2006

In this paper, we present the traditional GPS Position- Velocity (PV) model to apply for both Discrete-Time Kalman Filter and Discrete-Time $H_{\infty}$ Filter. The positioning algorithms of both filters are proposed for a stand-alone low-cost GPS module to increase its accuracy. For disturbance cancellation, the Kalman Filter requires the statistical information about process and measurement noises while the $H_{\infty}$ Filter only requires that these noises are bounded. Experiments show that with the same measurement data, $H_{\infty}$ Filter gives us better positioning results compared with Least-Squared method and Kalman Filter.

• 한국정보디스플레이학회:학술대회논문집
• /
• 2006.08a
• /
• pp.399-402
• /
• 2006

A new voltage-scaled compensation pixel which employs 3 p-type poly-Si TFTs and 2 capacitors without additional control line has been proposed and verified. The proposed pixel does not employ the $V_{TH}$ memorizing and cancellation, but scales down the inevitable $V_{TH}$ variation of poly-Si TFT. Also the troublesome narrow input range of $V_{DATA}$ is increased and the $V_{DD}$ supply voltage drop is suppressed. In our experimental results, the OLED current error is successfully compensated by easily controlling the proposed voltage scaling effects.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.149-152
• /
• 1999

This paper addresses major issues in localising sound sources in space using the experimental data set of head-related responses in the time or frequency domain. They come from the technical realisation steps for implementing the convolution of HRIR's with sound sources, the cross-talk cancellation for transaural filtering, the matched time delay compensation, etc. in real, those technical matters seem to be minor because they can be realised in off-line signal processing schemes. This paper puts much emphasis on what we misunderstood about the sets of HRTF's or HRIR's, More specifcaily, the sets of HRTF's or HRIR's of course supply relevant information to sound localisation but include much useless 'rubbish' that have made for us to fail to put spatial image into real souno signals such as voices and music's. This paper proposes possible reasons for such failure and, furthermore, introduces detained subjects that should be challenged so as to resolve them.

• The Journal of the Acoustical Society of Korea
• /
• v.19 no.8
• /
• pp.26-34
• /
• 2000

This paper presents a new adaptive impulse noise cancelling technique based on the adaptive nonlinear suppressing function. The proposed "adaptive threshold algorithm (ATA)" is controlled by the normalized power prior input data term, and this adaptive threshold makes the cancelling system highly robust against additive impulse noise. For the performance evaluation, we have tested the proposed algorithm with the observed signals simulated in various impulsive noise environments and real EMG signals. As a result the proposed algorithm shows superior performance of 51.7% to the available techniques in the points of SNR and MSE.

• Proceedings of the IEEK Conference
• /
• 2003.07e
• /
• pp.2471-2474
• /
• 2003

The modified discrete cosine transform (MDCT) and its inverse transform (IMDCT) are employed in subband/transform coding schemes as the analysis/synthesis filter bank based on time domain aliasing cancellation (TDAC). And they are the most computational intensive operations in layer III of the MPEG audio coding standard. In this paper, we propose a new efficient algorithm for the MDCT/IMDCT computation. It is based on the MDCT/IMDCT computation algorithm using the discrete cosine transforms (DCTs), and it employs two discrete cosine transform of type II(DCT-II) to compute the MDCT/IMDCT. In addition to, it takes advantage of ability in calculating the MDCT/IMDCT computation, where the length of a data block is divisible by 4. The proposed algorithm in this paper requires less calculation complexity than the existing methods. Also, it can be implemented by the parallel structure,, and its structure is particularly suitable for VLSI realization.

• Journal of Korea Society of Digital Industry and Information Management
• /
• v.8 no.4
• /
• pp.129-133
• /
• 2012

Coordinated multi-point transmission (CoMP) is a candidate technique for next generation cellular communications systems. One of the primary elements discussed in LTE-Advanced technology is CoMP, which can improve cell edge user data rate as well as spectral efficiency due to multiple input multiple output - orthogonal frequency division multiplex (MIMO-OFDM). We consider a system with multiple cells in which base stations coordinate with each other by sharing user channel state information (CSI), which mitigates inter cell interference (ICI), especially for users located at the cell edge. We introduce a new user scheduling method of ICI cancellation and the loss reduction of effective channel gain during the beamforming process, the proposed method improves the system sum rate, when compared to the conventional method by an average of 0.55bps/Hz in different number of total users per cell. It also outperforms the conventional method by approximately 0.38bps/Hz using different SNRs.